No audio for sip calls

Could please share your pjsip.conf

Obvious options in the type=aor section.

These are two separate options. comedia is something like symmetric_rtp, force rport should be obvious. However, if a provider tells you to use these, to access them, they either don’t understand them, and are doing a blind copy and paste, or they are so badly configured you really need a competent provider. This should only be needed if you have a complex network, or remote users.

This is two options, something like external_signalling and external_media, but check the documentation and samples.

The option for this should be obvious, but the parameter is wrong. It needs to be one or more network specifications, so each one needs to have both the network part of the address and the netmask. I believe both standard format for the netmask work (with either the number of bits, or the bit mask).

i have a no idea about the asterisk configurations i tried to setup it through the documentations but the problem i am facing is on some networks audio is ok but on some it is not if you can check this thread please Asterisk server audio issue for some networks - #3 by Rushikesh-Zealconnec